similar to: custom ring tone

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

Hello All! Well I still have users which almost daily get corrupt dovecot files in their IMAP folders (Maildir style) and are unable to read their mail in this particular folders. When I delete dovecot files in this folder (rm -rf dove*) user can again read all messages. This is regular Linux CentOS 4.2 with SAN storage (boot from SAN) as disks. So this is EXT3 on LVM. BUT it is VERY

I have asterisk with following users; a) zaphfc ISDN card with two channels b) two mediatrix FXS gateways with four channels each c) 1x CISCO 7905G d) two notebooks with MS Messenger 4.7 Now, it seems that any combination works correctly in all combinations except when I call from MS messenger and then call is dropped always in 25th second of the call. Any ideas what I did wrong? here is my

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

I have 2 Digium 4 port FXO cards in my system. The system is a P4 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1 for storage - whitebox - running RedHat 9. With pretty much any CVS HEAD version we are getting, what I will call, "phantom" calls on some lines. What I mean by a phantom call is that the line will ring, Asterisk will log that the Zap channel has been

I'm not aware of any routine that those the job, although I think that it could be relatively easily done by multiplication the manifest variable vector with the estimates for the specific effect. To make an example: v1; v2; v3; v4 are manifest variables that loads on one y latent variablein a data frame called "A" the code for the model should be like: model <-specifymodel( y

Here's a patch against 4.8-RELEASE kernel that allows disk writes on softupdates-enabled filesystems to be delayed for (theoretically) arbitrarily long periods of time. The motivation for such updating policy is surprisingly not purely suicidal - it can allow disks on laptops to spin down immediately after I/O operations and stay idle for longer periods of time, thus saving considerable amount

Updated from ports collection from ver. dovecot-1.0.a3 to 1.0.a4_1. And it wan't work. Here's the log Nov 8 10:51:07 ot-group dovecot: Dovecot v1.0.alpha4 starting up Nov 8 10:51:08 ot-group dovecot: Auth process died too early - shutting down Nov 8 10:51:08 ot-group dovecot: auth(default): Unknown userdb driver 'passdb' Nov 8 10:51:08 ot-group dovecot: child 17872 (auth)

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

hi how to generate the api documentation localy with the same frameset ...i mean to be abble to have files classes and methods listing on left. thanks

I?ve gotten PlayTones to work, however it stops playing the tones as soon as the call is answered. I would like to use PlayTones during the call because I want to have a tone/beep played in the background while call recording is going on. Anyone know a way to get PlayTones to work while call is in progress? Alternatively, does anyone have a suggestion for playing the tone/beep for recorded

Hy there I just saw some rails videos. *NICE!!!* So I got this basic principle question: Now I created my jawdropping application. But it''s just running localy. Can I kind of export my project to php&sql. so I can import the database and upload the php to my server to running it "outside"? Or do I have a provider that runs ruby? I dont get this very basic problem. All I

I have joined my win2k client machines (win2k professional or win2k professional server) to my PDC. This worked fine and I did all recommended as e.g. in the HOWTO http://hr.uoregon.edu/davidrl/samba/samba-pdc.html#joining. So far I can also login localy to the win2k client machines using a local account and connect a network drive from the PDC using a valid account on the PDC (as given in the

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

I have clang from 2010-01-22 installed. I'm interested in using the AST, but: marko at prpad:tmp% clang -cc1 col.c -ast-print gcc: unrecognized option '-cc1' /tmp/cc-15456a.s: Assembler messages: /tmp/cc-15456a.s:4: Warning: unterminated string; newline inserted /tmp/cc-15456a.s:5: Warning: unterminated string; newline inserted gcc: unrecognized option '-cc1' How can I use